1. Field of the Invention
The present invention relates to guaranteeing Quality of Service (QoS) in a Voice over Internet Protocol (VoIP). More particularly, the present invention relates to a method and system for guaranteeing QoS of a VoIP by reserving a bandwidth in a layer 2.
2. Description of the Related Art
QoS is a service level required for a specific application program. Since an Internet network based on an IP provides a best-effort data service, it does not guarantee a high QoS. However, it is necessary to guarantee QoS on the IP network in order to embody the VoIP.
In the conventional technology to guarantee the QoS, there are two protocols, that is, a Diffserv (Differentiated Services) protocol that uses priority queuing via various queuing methods in a layer 3 and a RSVE (Resource Reservation) protocol that guarantees a bandwidth.
Meanwhile, a layer 2 is a scheme used to guarantee the QoS, which introduces only a priority scheme utilizing IEEE 802.1p and does not introduce a QoS guarantee scheme via a guarantee of the bandwidth.
It is necessary to guarantee a sufficient bandwidth in order to guarantee an optimum quality of sound in the case of VoIP. However, a minimum bandwidth for the VoIP service is sometimes not guaranteed due to a bandwidth congestion phenomenon in a service in which the bandwidth is variably used by sharing a general data service and a bandwidth in the VoIP.
Such a problem does not occur when VoIP terminals exist in the same switch since the switch has a property of guaranteeing a wire speed between the VoIP terminals. However, such a problem sometimes occurs when the VoIP terminals exist in different switches since the switch has an uplink bandwidth limitation between the VoIP terminals.
In the case of using a hub, of course, such a problem also occurs between the VoIP terminals positioned in the same hub. In this case, the loss and delay of a VoIP packet increases due to a bandwidth loss, and the jitter increases due to the variable data service so that it has a serious effect on the VoIP sound quality.
In an LAN system, a backbone switch is connected to a number of switches, that is, a switch a, a switch b, a switch c, a switch d, . . . , a switch x. The backbone switch is also connected to a data terminal, such as a Personal Computer (PC).
The switch a and switch b are connected to various data terminals. These data terminals can include an IP phone, a PC, a PDA, and so on.
The switch a is also connected to other switches, a switch a′ and a switch b′, and the switch a′ is connected to a switch a″ and a switch b″.
One switch is connected to a number of switches and those switches are connected to other switches, and such a process can be continuously expanded.
When the uplink of the switch a supports a bandwidth of 100M and each of the switch a′ and switch b′ also has a bandwidth of 100M for their uplinks, a problem can occur when a large quantity of data is transmitted from a PC#5 connected to the switch a″ to a PC#1 connected to the switch a.
Since the uplinks of the switch a″ and switch a′ are limited to a bandwidth of 100M, respectively, a severe data congestion phenomenon occurs when a data transmission service is performed between terminals connected to the switch a and switch a′ and other terminals connected to the switch a″ or the switch b″.
This is due to a lack of bandwidth, which has a serious effect on a quality of sound when VoIP service is provided between an IP Phone #1 and an IP Phone #3.
In particular, a window size is changed for flow control for a data service that uses a TCP protocol, where the bandwidth is variably used, which causes a problem of increasing the jitter that has the most noticeable effect on a quality of sound in the VoIP service.